dograh/api/services/pipecat/event_handlers.py

245 lines
9.3 KiB
Python

from loguru import logger
from api.db import db_client
from api.services.campaign.call_dispatcher import campaign_call_dispatcher
from api.services.pipecat.audio_config import AudioConfig
from api.services.pipecat.audio_transcript_buffers import (
InMemoryAudioBuffer,
InMemoryTranscriptBuffer,
)
from api.services.pipecat.pipeline_metrics_aggregator import PipelineMetricsAggregator
from api.services.workflow.disposition_mapper import (
apply_disposition_mapping,
get_organization_id_from_workflow_run,
)
from api.services.workflow.pipecat_engine import PipecatEngine
from api.tasks.arq import enqueue_job
from api.tasks.function_names import FunctionNames
from pipecat.frames.frames import Frame
from pipecat.pipeline.task import PipelineTask
from pipecat.processors.audio.audio_buffer_processor import AudioBuffer
from pipecat.processors.audio.audio_synchronizer import AudioSynchronizer
def register_transport_event_handlers(
task: PipelineTask,
transport,
workflow_run_id,
engine: PipecatEngine,
audio_buffer: AudioBuffer,
audio_synchronizer: AudioSynchronizer,
audio_config=AudioConfig,
):
"""Register event handlers for transport events"""
# Initialize in-memory buffers with proper audio configuration
sample_rate = audio_config.pipeline_sample_rate if audio_config else 16000
num_channels = 1 # Pipeline audio is always mono
logger.debug(
f"Initializing audio buffer for workflow {workflow_run_id} "
f"with sample_rate={sample_rate}Hz, channels={num_channels}"
)
in_memory_audio_buffer = InMemoryAudioBuffer(
workflow_run_id=workflow_run_id,
sample_rate=sample_rate,
num_channels=num_channels,
)
in_memory_transcript_buffer = InMemoryTranscriptBuffer(workflow_run_id)
@transport.event_handler("on_client_connected")
async def on_client_connected(transport, participant):
logger.debug("In on_client_connected callback handler - initializing workflow")
await audio_buffer.start_recording()
if audio_synchronizer:
await audio_synchronizer.start_recording()
await engine.initialize()
@transport.event_handler("on_client_disconnected")
async def on_client_disconnected(transport, participant):
logger.debug("In on_client_disconnected callback handler")
await engine.handle_client_disconnected()
# Stop recordings
await audio_buffer.stop_recording()
if audio_synchronizer:
await audio_synchronizer.stop_recording()
# Cancel the task since the client is disconnected
await task.cancel()
# Return the buffers so they can be passed to other handlers
return in_memory_audio_buffer, in_memory_transcript_buffer
def register_task_event_handler(
workflow_run_id: int,
engine: PipecatEngine,
task: PipelineTask,
transport,
audio_buffer: AudioBuffer,
audio_synchronizer: AudioSynchronizer,
in_memory_audio_buffer: InMemoryAudioBuffer,
in_memory_transcript_buffer: InMemoryTranscriptBuffer,
pipeline_metrics_aggregator: PipelineMetricsAggregator,
):
@task.event_handler("on_pipeline_finished")
async def on_pipeline_finished(
task: PipelineTask,
frame: Frame,
):
logger.debug(f"In on_pipeline_finished callback handler")
workflow_run = await db_client.get_workflow_run_by_id(workflow_run_id)
# Stop recordings
await audio_buffer.stop_recording()
if audio_synchronizer:
await audio_synchronizer.stop_recording()
call_disposition = await engine.get_call_disposition()
logger.debug(f"call disposition in on_pipeline_finished: {call_disposition}")
gathered_context = await engine.get_gathered_context()
# also consider existing gathered context in workflow_run
gathered_context = {**gathered_context, **workflow_run.gathered_context}
organization_id = await get_organization_id_from_workflow_run(workflow_run_id)
mapped_call_disposition = await apply_disposition_mapping(
call_disposition, organization_id
)
gathered_context.update({"mapped_call_disposition": mapped_call_disposition})
# Set user_speech call tag
if in_memory_transcript_buffer:
call_tags = gathered_context.get("call_tags", [])
try:
has_user_speech = in_memory_transcript_buffer.contains_user_speech()
except Exception:
has_user_speech = False
if has_user_speech and "user_speech" not in call_tags:
call_tags.append("user_speech")
# Append any keys from gathered_context that start with 'tag_' to call_tags
for key in gathered_context:
if key.startswith("tag_") and key not in call_tags:
call_tags.append(gathered_context[key])
gathered_context["call_tags"] = call_tags
# Clean up engine resources (including voicemail detector)
await engine.cleanup()
# ------------------------------------------------------------------
# Close Smart-Turn WebSocket if the transport's analyzer supports it
# ------------------------------------------------------------------
try:
turn_analyzer = None
# Most transports store their params (with turn_analyzer) directly.
if hasattr(transport, "_params") and transport._params:
turn_analyzer = getattr(transport._params, "turn_analyzer", None)
# Fallback: some transports expose params through input() instance.
if turn_analyzer is None and hasattr(transport, "input"):
try:
input_transport = transport.input()
if input_transport and hasattr(input_transport, "_params"):
turn_analyzer = getattr(
input_transport._params, "turn_analyzer", None
)
except Exception:
pass
if turn_analyzer and hasattr(turn_analyzer, "close"):
await turn_analyzer.close()
logger.debug("Closed turn analyzer websocket")
except Exception as exc:
logger.warning(f"Failed to close Smart-Turn analyzer gracefully: {exc}")
usage_info = pipeline_metrics_aggregator.get_all_usage_metrics_serialized()
logger.debug(f"Usage metrics: {usage_info}")
await db_client.update_workflow_run(
run_id=workflow_run_id,
usage_info=usage_info,
gathered_context=gathered_context,
is_completed=True,
)
# Release concurrent slot for campaign calls
if workflow_run and workflow_run.campaign_id:
await campaign_call_dispatcher.release_call_slot(workflow_run_id)
# Write buffers to temp files and enqueue S3 upload
try:
# Only upload if buffers have content
if not in_memory_audio_buffer.is_empty:
audio_temp_path = await in_memory_audio_buffer.write_to_temp_file()
await enqueue_job(
FunctionNames.UPLOAD_AUDIO_TO_S3, workflow_run_id, audio_temp_path
)
else:
logger.debug("Audio buffer is empty, skipping upload")
if not in_memory_transcript_buffer.is_empty:
transcript_temp_path = (
await in_memory_transcript_buffer.write_to_temp_file()
)
await enqueue_job(
FunctionNames.UPLOAD_TRANSCRIPT_TO_S3,
workflow_run_id,
transcript_temp_path,
)
else:
logger.debug("Transcript buffer is empty, skipping upload")
except Exception as e:
logger.error(f"Error preparing buffers for S3 upload: {e}", exc_info=True)
await enqueue_job(FunctionNames.CALCULATE_WORKFLOW_RUN_COST, workflow_run_id)
await enqueue_job(
FunctionNames.RUN_INTEGRATIONS_POST_WORKFLOW_RUN, workflow_run_id
)
def register_audio_data_handler(
audio_synchronizer, workflow_run_id, in_memory_buffer: InMemoryAudioBuffer
):
"""Register event handler for audio data"""
logger.info(f"Registering audio data handler for workflow run {workflow_run_id}")
@audio_synchronizer.event_handler("on_merged_audio")
async def on_merged_audio(_, pcm, sample_rate, num_channels):
if not pcm:
return
# Use in-memory buffer
try:
await in_memory_buffer.append(pcm)
except MemoryError as e:
logger.error(f"Memory buffer full: {e}")
# Could implement overflow to disk here if needed
def register_transcript_handler(
transcript, workflow_run_id, in_memory_buffer: InMemoryTranscriptBuffer
):
"""Register event handler for transcript updates"""
@transcript.event_handler("on_transcript_update")
async def on_transcript_update(processor, frame):
transcript_text = ""
for msg in frame.messages:
timestamp = f"[{msg.timestamp}] " if msg.timestamp else ""
line = f"{timestamp}{msg.role}: {msg.content}\n"
transcript_text += line
# Use in-memory buffer
await in_memory_buffer.append(transcript_text)